Audio system with centralized audio signal processing

ABSTRACT

One or more audio conferencing systems are connected to a local network, and each conferencing system is comprised of a plurality of wireless microphones in communication with a plurality of antennas deployed in an array configuration. Each of the antennas comprising one of the audio conferencing systems is in direct communication with a base station and in indirect communication with a server which runs a centralized digital signal processing functionality. The digital signal processing functionality operates on audio information received from one or more far-end audio sources and from each of the one or more audio conferencing system.

FIELD OF THE INVENTION

The present disclosure relates to audio conferencing systems havingwireless microphones associated with an array of antennas linked to acommon network.

BACKGROUND

Meetings held in large rooms involving two or more individuals can befacilitated using a room audio system, or, in the case that individualsare conducting a remote meeting, an audio conferencing system can beused. Room audio systems or audio conferencing systems typically includesome number of microphones, at least one loudspeaker and a base stationwhich may or may not be linked to a network. In a room audio system,microphones can operate to pick up acoustic audio signals (speech) andtransmit the signals to a base station which generally operates toprovide session control and to process the audio signals in a number ofways before sending it to a loudspeaker located in the room to beplayed. Among other things, the base station can be configured toamplify audio signals, it can regulate signal gain and suppress noise,and it can remove acoustic echo from signals received by themicrophones.

FIG. 1 is a diagram showing functional elements comprising acommercially available room audio system 100. The system 100 can becomprised of a number of wireless microphones 11 and/or wiredmicrophones 12, one or more loudspeakers 13, and an audio control andprocessing device 14. Typically, in such room audio systems, theloudspeakers 13 are wired to the device 14 and positioned within theroom so that all of the individuals in the room can easily hear whateach of the other individuals in the room is saying, and the processingdevice 14 includes complex digital signal processing and audio signalcontrol functionality. Depending upon its capability, the processingdevice 14 can be a relatively expensive element of the overall systemcost. Including a separate device 14 in each room in which an audiosystem is installed can be expensive.

A commercially available audio conferencing system 20 illustrated inFIG. 2 is comprised of some number of wired or wireless microphones 21and loudspeakers 22 associated with an audio control and processingdevice 23 (such as a base station) which is in communication with one ormore audio sources (can be local or remote, far end sources) over eithera local or a wide area network. In addition to having audio sessioncontrol functionality such as audio channel mixing, amplification andgain control, the device 23 can also have functionality for removingfeedback and acoustic echo from the microphone signals. High qualityaudio is facilitated in such an audio conferencing system havingmultiple, wireless microphones attached to an individual which allowsthe individual to move around a room with the microphone during aconferencing session. As with conferencing system 100 described earlierwith reference to FIG. 1, a separate processing device 23 is included ineach room in which the system is installed and is typically the mostexpensive part of the system.

Another audio conferencing system 300 configuration is shown withreference to FIG. 3 in which each of a plurality of separate conferencephones (31A-31Z) have an integrated microphone, loudspeaker, control andhandset. Each of the phones are shown to be connected over a localnetwork 32 to a common audio processing device 33. In thisconfiguration, all of the audio signal processing is performed by adigital signal processor (DSP) 34 comprising the device 33. In order fora system in this configuration to perform acoustic echo cancellation(AEC), it is necessary that the timing of an audio sampling function ateach of the phones and at the DSP 34 be tightly correlated orsynchronized. Having the signal processing functionality located in asingle, network device as opposed to including this functionality ineach of the conference phone simplifies the operation and lowers thecost of each phone. While the system 300 in FIG. 3 is reasonably priced,it is not well suited for a large meeting room or conference roomapplication in which there are many individuals spread out in the roomor in which the individuals are moving around the room.

BRIEF DESCRIPTION OF THE DRAWINGS

The present invention can be best understood by reading thespecification with reference to the following figures, in which:

FIG. 1 is a diagram showing the functional elements of a room audiosystem 100.

FIG. 2 is a diagram showing the functional element of an audioconferencing system 200.

FIG. 3 is a diagram showing an audio conferencing system 300 havingseparate conference phones linked to an audio processing device.

FIG. 4 is a diagram of an embodiment of an audio conferencing system 400with a centralized audio processor and an array of antenna/transceiverdevices.

FIG. 5 is a diagram showing functionality comprising an antenna array500.

FIG. 6 is a diagram showing functionality comprising a base station 45A.

FIG. 7 is a diagram showing functionality comprising a server 50.

FIG. 8 is a diagram showing modules comprising a time synchronizationfunction 800.

DETAILED DESCRIPTION

Cross Reference to Related Applications: This application claims thebenefit under 35 U.S.C. 120 of U.S. patent application Ser. No.13/541,148 entitled, “Synchronizing Audio Signal Sampling in a WirelessDigital Audio Conferencing System”, filed Jul. 3, 2012, the entirecontents of which are incorporated herein by reference.

Typically, a room audio system or audio conferencing system usingwireless microphones has a single base station positioned in a room withthe microphones to receive signals from and send signals to each of themicrophones. The base station can include a transceiver or radiodesigned to operate according to a particular wireless protocol such asWiFi or DECT and it can have audio control and audio signal processingfunctionality. A room audio system is generally self contained and notlinked to a network while an audio conferencing system is typicallylinked to either a local network such as an Ethernet, or it is linked toa wide area network such as the Internet.

While configuring each system with a base station proximate to thewireless microphones simplifies audio signal processing, it addsunnecessarily to the overall cost of the system, and restricts thesystems use to the room in which the system is installed. It wasdiscovered, according to one embodiment, that it is possible to designand deploy a low cost, flexible/mobile/transportable room audio or audioconferencing system having a single, central audio processing device(DSP) which is in communication with two or more antenna groups, eachantenna group being geographically separated from each other antennagroup, and each antenna group having at least two antennas. One or morewireless microphones can be associated with each antenna in an antennagroup, and each antenna can be wirelessly linked to the centralizedaudio processing device (i.e, base station) so that one or more antennain an antenna group can be easily transported between meeting rooms thatare within range of the central audio processing device.

According to another embodiment, a single, central audio processingdevice is connected to two or more base stations, and each base stationis connected to two or more geographically separated antenna groups,with each antenna group having at least two antennas. In order to cancelecho from an acoustic audio signal received at any of the microphones,the timing of audio sampling functionality at the microphones and audiosampling functionality at the centralized audio processing device istightly synchronized to facilitate the operation of the digital signalprocessing functionality.

According to each embodiment of the invention described above, the twoor more geographically separated antenna groups comprise a singleantenna array that is connected, either wirelessly or is wired to, asingle base station. Depending upon the manner in which the antennaarray is physically configured, each of the wireless microphonesassociated with an antenna in an antenna group in the antenna array canroam from one antenna to another antenna, and roam from one antennagroup to another antenna group without interrupting an audio signal toor from a microphone.

A local network 400 shown in FIG. 4 is comprised of at least one networkswitch 49, a computational device such as a server 50, and one or moreaudio systems 41A and 41B. According to this description, the audiosystem 41A and 41B can be a room audio system or an audio conferencingsystem that is connected to a public network, such as the Internet. Eachaudio system 41A and 41B utilizes a central digital signal processingfunction running on the server 50 to perform acoustic echo cancellationon an audio signal received at any of a plurality of microphones 57comprising each of the audio systems 41A and 41B. Each audio system 41Aand 41B in FIG. 4 comprises a single base station 45A and 45Brespectively, a plurality of antenna/receiver devices 56A and 56B, aplurality of wireless microphones 57A and 57B, one or more associatedloudspeakers (either wired or wireless) 58A and 58B, and an optionalwireless control module 59A and 59B respectively. Each of themicrophones 57A and 57B, antennas 56A and 56B, loudspeakers 58A and 58B,and the optional control module 59A, are shown installed in two separaterooms 42A and 42B respectively. The antennas 57A installed in room 42Aare referred to as an antenna group, with a different antenna groupinstalled in each room 42A and 42B. Together, the antenna groups inrooms 42A and 42B comprise an antenna array, and the antenna array isconnected to a single base station 45A in FIG. 4. The base station is inthis case connected to the network switch 49 which is, in turn connectedto a public network 60. The switch 49 is, in this case, connected to twoaudio systems 41A and 41B and to the server 50. The server 50 hasdigital signal processing functionality that is specially designed tocancel acoustic echo in audio signals received at any of the microphones57B and 57B.

Referring to FIG. 4 in more detail, room 42A has one or more antennas56A each of which is associated with one or more of the wirelessmicrophones 57A, at least one of the loudspeakers (wired or wireless)58A, and the optional wireless control device 59A. Each room 42A and 42Bin the audio system 41A can have a similar number and types of systemcomponents or they can have a different number and types of systemcomponents. Each room can have the same number of antennas or not, eachroom can have the same number of microphones or not, each room can havethe same number of loudspeakers or not, and each room can have a systemcontrol device or not. The similarity and differences between thecomponent parts comprising an audio system in each room typicallydepends upon the size of the room, the number of individuals using thesystem, and any user specified requirements, such as an individual'smobility.

Continuing to refer to FIG. 4, depending upon the audio conferencingsystem components installed in each room, each of the base stations 45Aand 45B supporting the audio systems 41A and 41B respectively can havesimilar or different audio control functionality, but generally eachbase station can have functionality to control the mixing of two or moreaudio channels and to control the gain of each audio channel. The switch49 generally operates to receive frames of audio and other informationfrom the base stations 45A and 45B, from the server 50, and to receivepackets comprising audio and other information from a far-end audiosource and forwarding this information to a destination within network400. The server 50 is comprised of, among other things, digital signalprocessing functionality, described later with reference to FIG. 6, thatcan operate to cancel acoustic echo from samples of audio informationcaptured by any of the microphones 57A and 57B associated with the eachaudio system. In order for either of the audio systems 41A or 41B tooperate to cancel acoustic echo from each of the microphone signals, itis necessary to synchronize the timing of the audio sampling functionsrunning on all of the wireless microphones and running in associationwith the digital signal processing (DSP) on the server 50.Synchronization in the context of this description relates to the audiosystem time at which samples of audio information are captured at amicrophone and at a base station such that substantially the same audioinformation is captured in both samples, albeit the sample captured atthe microphone represents audio information in a format that istransmitted over a wireless medium, while the sample captured at thebase station represents audio information in a format that istransmitted over a wired medium. Finally, the control module 59 is anoptional element that can be employed by audio system users to amongother things initiate and terminate conferencing sessions, to mutenear-end audio, and to control loudspeaker gain.

In order to effect the audio system timing synchronization describedabove, it is necessary for each microphone, each antenna, each basestation, the switch and the server comprising the network 400 in FIG. 4to implement a timing synchronization protocol. The same protocol can beimplemented in each device comprising the audio system, or more than oneprotocol can be strategically implemented. One such timingsynchronization protocol is described in two IEEE standards documents,the first one of which is entitled “Local and Metropolitan AreaNetworks—Audio Video Bridging (AVB) Systems” (IEEE Std 802.1BA-2011) andpublished in 30 Sep. 2011, and the second one of which is anInternet-Draft standard entitled “Synchronization for RTP Streamsdraft-Williams-avtext-avbsync-01 (IEEE 1588/802.1AS). Both of these IEEEdocuments are publically available and the entire contents of thesedocuments are incorporated herein by reference. Another timing protocolis described in the earlier referenced U.S. patent application.

Continuing to refer to FIG. 4, and in another embodiment, each of theaudio systems 41A and 41B described with reference to FIG. 4 may notinclude the base stations 45A and 45B respectively. In thisconfiguration, the switch 49 operates to forward packets/frames of audioinformation to the appropriate destination antenna group located in eachof the different rooms and to receive audio information from each of theantenna groups in the different rooms. Further, the functionalitycomprising the base stations can be implemented in the digital signalprocessing functionality associated with the server 50, for example. Thegeneral operation of the audio conferencing system 41A of FIG. 4 tosupport an audio session is described below.

In operation, the network switch 49 in FIG. 4 receives packets of audioinformation transmitted to it by a far-end audio source, such as anotheraudio system. The switch 49 can forward the audio information comprisingthe packets to the server 50 where digital signal processingfunctionality can capture a first sample of audio information and storethis first sample for later use. The server 50 can then send the audioinformation back to the switch 49 which can in turn forward the audioinformation to the base station 45A. The base station 45A can thenadjust the gain and/or mix the audio information with other audioinformation and send this mixed audio information to a loudspeaker, suchas loudspeaker 58A in room R42A, to be played. One or some of thewireless microphones 57A in room 42A can receive acoustic audioinformation played by the loudspeaker, and it can receive acoustic audioinformation from another audio source in the room, such as from one ormore individuals speaking in the room. The acoustic audio informationreceived by the microphone is captured in a second sample of audioinformation and sent to the server 50 for processing. The DSPfunctionality running on the server then uses the audio informationcomprising the first sample to remove acoustic echo in the second sampleof audio information, and the server can send the audio information withthe echo removed back to the switch for transmission to the far-end.

In an alternative embodiment the digital signal processing functionalityrunning on the server 50 can be located in each of the base stations,but otherwise the configuration and operation of the conferencingsystems in FIG. 4 is the same. A more detailed description of thecomponent parts of the audio conferencing system 41A of FIG. 4 isundertaken below with reference to FIG. 5 and FIG. 6.

The audio conferencing systems 41A and 41B described earlier withreference to FIG. 4 each comprise a plurality of antennas 56B and 56Bwhich are deployed in association with each audio system in one or moregeographically separate locations. According to this embodiment,geographically separate locations can be construed to mean that aplurality of antennas (an antenna group) are deployed in one room butseparated from each other spatially, that a plurality of antennas aredeployed in each of two or more different rooms, or that a plurality ofantennas are deployed in each of two or more different buildings. Theantenna deployment arrangement associated with each audio system isreferred to here as an antenna array. Unless an audio system isinstalled in only one room, each separate location in which a group ofantennas is installed comprises a portion of the antenna array, and eachportion (group) can include one or more antennas. Such an antenna array500 is now described below with reference to FIG. 5.

As described above, each antenna array is comprised of two or moreantennas deployed in antenna groups at one or more geographic locations.Each antenna in the array can be connected to a base station, such asthe base station 45A of FIG. 4, over either a wired or wireless networklink, and each antenna in the array can be connected to one or morewireless microphones (not shown) over a wireless link. FIG. 5 shows aplurality of antenna groups 56A, 56B to 56Z each group deployed in oneof a plurality of locations or rooms 42A to 42Z respectively. Thedistribution between wireless and wired network links can vary fromlocation to location in the antenna array depending upon the applicationfor which the audio system is used. One location can have antennas thatare all connected to a base station over a wireless link, have antennasconnected to the base station over wired links, or have some combinationof antennas with wired and wireless links to the base station.

Continuing to refer to FIG. 5, each antenna 56A has a transceiver (mic.radio) for communicating with one or more microphones, and each antennahas an interface for communicating with the base station 45A. Each mic.radio can be designed to support the transmission or reception of two ormore channels of audio information in a time division multiplexedmanner. Each channel can be dedicated to communication with a particularone of a plurality of microphones 57A in FIG. 4. Antennas connected tothe base station 45A over a wireless link, such as antenna one of theantennas 56A, has a wireless transceiver such as an 802.11 radio,whereas antennas connected to the base station 45A over a wired linkhave a wired interface such as an Ethernet interface card (NIC). Asdescribed earlier, all of the antennas comprising the audio system 41Ahave a transceiver in order to communicate with a wireless microphone,and this transceiver can be a digital radio (DECT, WiFi or otherwireless technology) or it can be an analog radio. Regardless of themeans they employ to communicate with a base station, each of theantennas can run a time synchronization protocol 510 such as thatreferred to earlier in the two IEEE publications. Generally, the timesynchronization protocol is implemented in each device in the network400 of FIG. 4 along a path between a microphone and a device having theDSP. This time synchronization protocol generally operates todetect/compute signal delay over links (wired or wireless) between anantenna on which the protocol is running and an associated wirelessmicrophone and base station. Specifically, the time synchronizationprotocol operates to calculate a signal propagation delay over a link(such as a link between a base station and an antenna) and it operatesto calculate a signal residence time delay (which is the time that ittakes a signal to pass through an antenna).

FIG. 6 is a block diagram illustrating the functional elementscomprising a base station such as the base station 45A of FIG. 4. Thebase station 45A has either a radio (digital or analog) or a physicalnetwork interface device 601 for communicating with each antenna in anantenna array, such as the array 500, and it has a physical networkinterface 602 connected over to a link 605 to the network switch 49 ofFIG. 4. The base station 45A can also have audio signal controlfunctionality 611 and means for running a time synchronization protocolfunctionality 612. These means can be a special or general purposecomputational device operating in conjunction with a non-volatile memorydevice or the means can be either of the two interface devices 601 and602. Regardless, the time synchronization protocol functionality 612 canoperate in a manner similar to, and be the same as or compatible with,the time synchronization functionality 510 described earlier withreference to FIG. 5. The base station 45A generally operates to receiveone or more channels of audio information over the link 605 at thenetwork interface 602, use the audio control function(s) 611 to mix twoor more of the channels of audio information and to regulate the gain ineach channel, and use the radio 601 to transmit/send the audio signal toan antenna. Depending upon the number of audio systems comprising thenetwork 400 of FIG. 4, the base station 45A can be connected over thenetwork link 605 directly to a network switch, such as the switch 49 inFIG. 4, which variously operates to forward audio information to anaudio system or to receive audio information from an audio system, orthe base station 45A can be connected directly to a server (in the casethat a network includes only one conferencing system), such as theserver 50 in FIG. 4, running acoustic echo cancellation functionality.The operation of the network switch 49 is not described here in anydetail other than mentioning that it does implement the timesynchronization functionality described earlier with reference to theantenna(s) in FIG. 5.

Functional elements comprising the server 50 are now discussed in moredetail with reference to FIG. 7. The server 50 is connected to thenetwork 400, via a NIC 700, such that it can receive audio informationfrom a far-end audio source and such that it is addressable by any ofthe audio systems comprising the network. As previously described,although acoustic echo cancellation (AEC) functionality is describedherein as running in a network server, the AEC functionality can beimplemented on any suitable computational device which can be configuredto operate in a network environment and which can be configured with adigital signal processing device on which the AEC functionality can run.Accordingly, the server 50 is comprised of a central processing unit(CPU) 701, a non-volatile/non-transitory memory device 702, one or moredigital signal processors 703 and associated AEC functionality 704, andtiming synchronization protocol functionality 705 stored in memory 702.Depending upon the number of audio systems comprising the network 400,the complexity of each of the audio systems and the capability of a DSP,one or more than one DSPs 703 may be necessary the systems echocancellation requirements. Regardless, each DSP 703 has AECfunctionality 704 that generally operates to receive, sample and storefar-end audio information, to receive samples of audio information fromeach microphone, and to remove the component of the audio informationreceived from the microphones that constitutes acoustic echo (which isequivalent to an estimate of the sampled far-end audio informationplayed over a loudspeaker). The time synchronization protocolfunctionality 705 can comprise software code which is stored in memory702 where it can be operated on by the CPU 701. The process by whichacoustic echo is removed from a near-end microphone signal is notdescribed here in any detail, as AEC processes are well known to thoseskilled in the art.

Prior art audio conferencing systems are design such that the AECfunctionality is as close to the system microphones as practical(typically in the same room). Proximity of the AEC functionality to themicrophones simplifies the methods employed to remove acoustic echo fromthe microphone signals. Specifically, in order to effectively cancelacoustic echo from audio samples captured at a microphone, it isessential that the system is able to synchronize/correlate the time atwhich the AEC functionality 704 in server 50 samples a far-end audiosignal with the sampling time of an acoustic signal at any of themicrophones 57A for instance. Without this correlation in time betweenthe two sampling functions, it is not possible to remove an acousticecho component from a microphone signal. The farther the AECfunctionality is moved away from the microphones in an audio system, themore difficult it becomes to correlate the timing of the two samplingfunctions. In order to overcome this problem, the time synchronizationfunctionality described earlier can be employed to very accuratelyprovide the sample time correlation information necessary to run an AECfunction that is positioned in a centralized network location, remotelyfrom the microphones comprising an audio system.

The functional elements comprising the time synchronization protocolimplemented on the various devices in the network 400 are described withreference to a published IEEE document entitled “802.1AS Tutorial” dated2008, Nov. 13 and authored by Kevin B. Stanton. This IEEE documentdescribes in detail how to implement the time synchronization protocoland so it will not be described here in any detail. However, the signaldelays that should be calculated when implementing the conferencingsystem 400 described with reference to FIG. 4 is described below withreference to FIG. 8.

FIG. 8 is an illustration of the functional elements/devices comprisingnetwork 400 that can be Included in the audio conferencing system 41A ofFIG. 4. As previously described, in order to remove acoustic echo from amicrophone signal, it is necessary to correlate the sample times of botha far-end audio signal received by the system 41A and the sample time ofan acoustic signal received at a microphone. In order to correlate thetime of the samples, it is necessary to calculate (using the timesynchronization protocol) all of the delays in the signal path betweenthe server 50 (running the DSP which implements the AEC) and thewireless microphones 57A for instance. Specifically, it is necessary tocalculate all of the delays in the network signal path from the locationat which the far-end audio signal is sampled, which is the DSP in thiscase, to the location at which an acoustic signal corresponding to thefar-end audio signal played by the loudspeaker 58A is sampled, which isone of the wireless microphones 57A in this case. This path signal delaycalculation is made as follows.

After the network selects a master clock to which at least some of thenetwork device timing can be synchronized, the network device in whichthe master clock is running can generate a first and a second mastertiming frame that is propagated to the devices connected (wirelessly orwired) to the network, which can be network 400 in this case. The firstand the second master timing frames can include, among other things, atime stamp corresponding to the time (T1) at which the frame istransmitted by the network device, which in this case can be the server50 of FIG. 4, and the master timing frames can be transmittedperiodically according to a time interval selected by a networkadministrator for instance. For the purposes of this description, it isassumed that one network path between the server 50 and a microphone 57Ais the path that traverses the server 50, the switch 49, the basestation 45A, an antenna 56, a loudspeaker 58A and/or a microphone 57A.Each instance of a time synchronization function located at each networkdevice along the path uses the time stamp information included in thefirst master timing frame to calculate a link delay and a residencetime. These times are temporarily stored by the network devices at leastuntil the second master timing frame is received by the network device.The second master timing frame incorporates the delay and residence timecalculated at each network device in the path into one or more fieldscomprising the timing frame as delay and resident time information, andwhen the timing frame reaches its destination, which in this case is themicrophone 57A, all of the delays and residence times can be summed bythe timing synchronization protocol running on the microphones whichresults in a time T1 plus a total network signal path delay time value(PathDelayTotal). The value of the PathDelayTotal can be used by themicrophone audio sampling functionality to correlate samples of audioinformation captured at the microphone with audio information sampled atthe DSP running in the server 50.

Returning to FIG. 8, two master timing messages M1 and M2 aretransmitted by the server 50 and propagate over the network pathstarting at the server 50, through the switch 49, the base station 45A,an antenna 56A, to the loudspeaker 58A and/or the wireless microphone57A. As previously described, the message M1 comprises time stampinformation relating to the time the message is transmitted by theserver 50. This time stamp information is inserted into the message by aserver output port. Each of the time synchronization functions locatedin the network devices in the path use the time stamp information in themessage M1 in order to calculate link delay between devices, and thesame time synchronization functionality is used to measure the residencetime for the time frame in the network device. The second master timingmessage M2 is transmitted a selected time subsequent to the first mastertiming message M1, and this message is operated on by the timingsynchronization function in each network device in the path to place thelink delays and the residence time into fields comprising the messageM2. When the message M2 arrives at its destination, which in this caseis the wireless microphone 57A, all of the link delay and residence timeinformation is included in one or more fields comprising the message andas described above is used to determine the total signal delay betweenthe server and the microphone.

The forgoing description, for purposes of explanation, used specificnomenclature to provide a thorough understanding of the invention.However, it will be apparent to one skilled in the art that specificdetails are not required in order to practice the invention. Thus, theforgoing descriptions of specific embodiments of the invention arepresented for purposes of illustration and description. They are notintended to be exhaustive or to limit the invention to the precise formsdisclosed; obviously, many modifications and variations are possible inview of the above teachings. The embodiments were chosen and describedin order to best explain the principles of the invention and itspractical applications, they thereby enable others skilled in the art tobest utilize the invention and various embodiments with variousmodifications as are suited to the particular use contemplated. It isintended that the following claims and their equivalents define thescope of the invention.

We claim:
 1. A wireless audio system, comprising: a plurality ofgeographically distributed wireless microphone groups each group havingat least one wireless microphone and being associated with at least oneof a plurality of loudspeakers; at least one antenna comprising each ofa plurality of geographically distributed antenna groups, the pluralityof the geographically distributed antenna groups comprising an antennaarray, and each one of the antenna groups in communication with one ofthe wireless microphone groups; a base station in communication witheach one of the plurality of the geographically distributed antennagroups, the base station having digital signal processing functionality; and each of the wireless microphones comprising each of the pluralityof geographically distributed wireless microphone groups, each of theantennas comprising each of the plurality of geographically distributedantenna groups, and the base station operate to run a timesynchronization protocol that correlates the time at which the basestation captures a first sample of audio information and the time atwhich one of the microphones captures a second sample of audioinformation; wherein the first and second samples of audio informationare used by the digital signal processing functionality to removesubstantially all of the acoustic echo associated with the second sampleof audio information.
 2. The system of claim 1, wherein the first andsecond samples of audio information are substantially the same.
 3. Thesystem of claim 2, wherein the first sample of audio informationcomprises information captured from the audio signal transmitted over awired medium and the second sample of audio information comprisesinformation captured from the audio signal transmitted over a wirelessmedium.
 4. The system of claim 3, wherein the second sample of audioinformation comprises an acoustic echo component that is cancelled bythe digital signal processing functionality at the base station.
 5. Thesystem of claim 1, further comprising a device for wirelesslycontrolling the operation of the audio system.
 6. The system of claim 1,further comprising any one of the wireless microphones roaming from oneof the plurality of the geographically distributed wireless microphonegroups to another one of the plurality of the geographically distributedwireless microphone groups.
 7. The system of claim 1, wherein each ofthe antennas have at least one receiver device for receiving signalshaving audio information from one of the wireless microphones.
 8. Thesystem of claim 1, wherein each wireless microphone is associated withone of an audio signal reproduction device (earphone).
 9. The system ofclaim 1, wherein the digital signal processing functionality operates tocancel acoustic echo in the second sample of audio information receivedfrom any one of the wireless microphones.
 10. A wireless audio system,comprising: two or more sets of a plurality of geographicallydistributed wireless microphone groups, each of the wireless microphonegroups having at least one wireless microphone and each of the wirelessmicrophone groups being associated with at least one of a plurality ofloudspeakers; two or more antenna arrays each comprising one set of theplurality of geographically distributed antenna groups, each one of theantenna groups having at least one antenna which is in communicationwith one of the plurality of geographically distributed wirelessmicrophone groups; and a digital signal processor located remotely fromand in communication with each one of the two or more antenna arrays,the digital signal processor having functionality that operates toremove acoustic echo from audio signals received from each of the two ormore antenna arrays.
 11. The system of claim 10, further comprising thewireless audio system having functionality that operates to run a timesynchronization protocol that correlates the time at which the digitalsignal processing functionality captures a first sample of audioinformation and the time at which one of the microphones captures asecond sample of audio information.
 12. The system of claim 10, whereinthe first and second samples of audio information are substantially thesame.
 13. The system of claim 11, wherein the first sample of audioinformation comprises information captured from the audio signaltransmitted over a wired medium and the second sample of audioinformation comprises information captured from the audio signaltransmitted over a wireless medium.
 14. The system of claim 13, whereinthe second sample of audio information comprises an acoustic echocomponent that is cancelled by the digital signal processingfunctionality at the base station.
 15. The system of claim 10, furthercomprising a device for wireles sly controlling the operation of theaudio system.
 16. The system of claim 10, wherein each of the antennashave at least one receiver device for receiving signals having audioinformation from one of the wireless microphones.
 17. The system ofclaim 10, wherein each wireless microphone is associated with one of anaudio signal reproduction device.
 18. The system of claim 10, whereinthe digital signal processing functionality operates to cancel acousticecho in the second sample of audio information received from any one ofthe wireless microphones.
 19. The system of claim 10, further comprisingeach of the two or more base stations are in communication with afar-end audio system over a public network.
 20. A method of processingan audio signal in a wireless audio system, comprising: capturing, at adigital signal processor, a first sample of audio information from theaudio signal at a first system time, and transmitting the audio signalto at least one of two or more of an antenna array, each antenna arrayhaving a plurality of geographically distributed antenna groups and eachantenna group having at least one antenna, the digital signal processorbeing located remotely from the two or more antenna arrays; transmittingthe audio signal from the at least one of the two or more of the antennaarrays to a loudspeaker associated with at least one of the antennagroups, and the loudspeaker playing the audio signal; receiving andcapturing, at a wireless microphone associated with the at least one ofthe antenna groups, at a second system time a second sample of audioinformation in an acoustic audio signal that substantially correspondsto audio information captured from the audio signal in the first sampleof audio information at the first system time, and sending the secondsample of audio information to the digital signal processor, and usingaudio information derived from the first sample of audio informationcaptured at the first system time to cancel substantially all of theaudio information comprising the second sample of audio informationcaptured at the second system time.
 21. The method of claim 20, furthercomprising the wireless audio system running a time synchronizationprotocol that correlates the time at which the digital signal processorcaptures a first sample of audio information and the time at which oneof the wireless microphones captures a second sample of audioinformation.
 22. The method of claim 20, wherein the digital signalprocessor comprises a computational device that is in communication witheach of the two or more of the antenna arrays.
 23. The method of claim22, further comprising the digital signal processor being incommunication with a public or a private network.
 24. The method ofclaim 20, wherein the audio signal is received by the wireless audiosystem from a public network or the audio signal is generated by anaudio source and captured by the wireless audio system.
 25. The methodof claim 20, wherein the first sample of audio information comprisesinformation captured from the audio signal transmitted over a wiredmedium and the second sample of audio information comprises informationcaptured from the audio signal transmitted over a wireless medium. 26.The method of claim 20, wherein the audio information that is cancelledis audio information associated with acoustic echo.